This present invention relates to a system and method for providing real-time data communications over packet switched networks. In particular, it relates to a system and method for compensating for channel jitter to provide a synchronous play-out of data packets.
Real time communications such as audio or video can be encoded using various compression techniques. One or more frames of encoded information can then be placed in a data packet with time and sequence information and transported via non-guaranteed Quality of to Service (QoS) packet networks. A stream of such packets may be used to transfer real time audio or video information. Non-guaranteed packet switched networks include a Local Area Network (LAN), Internet Protocol Network, Frame Relay Network, or an interconnected mixture of such networks such as an Internet or Intranet. One underlying problem with non-guaranteed packet networks is that transported packets are subject to varying loss and delays. Therefore, for real-time communications, a tradeoff exists among the quality of the service, the interactive delay, and the utilized bandwidth. This tradeoff is a function of the selected coding scheme, the packetization scheme, the redundancy of information packeted within the packets, the receiver buffer size, the bandwidth restrictions, and the transporting characteristics of the transporting network.
The transmission of real time communications over packet switched networks presents several challenges. One such problem is how to maintain a synchronous play-out of frames to the decoder, given that the packet arrival times are non-deterministic. The uncertainty in the arrival time of any packet in a packet stream is referred to as jitter. Typically, this problem is addressed using a jitter buffer. The jitter buffer waits for the reception of some number of frames before play-out begins. The threshold number of frames in the buffer during transmission is typically known as a watermark. In this way, if subsequent packets are slightly early or slightly late, they are still likely to arrive in time for their scheduled play-out.
The workings of the jitter buffer itself present different challenges. For example, if the buffer is made very large, the inherent delay of the jitter buffer may exceed the channel jitter, and the system receives all packets in time for their respective play-out. The disadvantage of having a very large buffer is that a large delay is added to the system. This may be unacceptable in a real time voice or video session. Conversely, if the buffer is made very small, there will be an excessive number of late packets which cannot be played on time. This will degrade the quality of the reconstructed signal. Thus, there is a trade-off between the buffer size and the quality of the reconstructed signal.
High quality switched networks are currently being designed to provide jitter within a certain range, such as, for example, xc2x140 ms. Theoretically, the buffer can be set to provide the precise amount of jitter compensation required, without introducing additional delay. However, locking on to a precise amount of jitter compensation is challenging. There is, therefore, still a need for providing a specified amount of jitter compensation without overcompensation or sacrificing the quality of the reconstructed signals.
In accordance with preferred embodiments of the present invention, some of the problems associated with transmission of real time data over packet switched networks are overcome. The present invention includes a system and method for a jitter buffer to maintain a specified value of jitter compensation. The method of the present invention includes receiving data packets at a network, followed by the header of the data packets being stripped off at a de-packetizer. The payload output from the de-packetizer includes one or more frames and a sequence number. The method also comprises buffering the data frames at a jitter buffer and shifting the play-out of the data packets from the jitter buffer to a decoder by at least a fraction of a frame to reduce delay between arrival times of the data packets and play-out time of the data packets.
In a particular embodiment the shift in the play-out of the data packets is calculated using a watermark value which represents the average number of frames in the buffer during steady state frame transmission. A first clock that defines the arrival sampling times runs at a rate faster than a second clock that defines the frame depletion times.
In another preferred embodiment the shift in the play-out of the data packets addresses a frame phase offset which occurs when the packet arrival intervals are out of phase with packet depletion times. The calculation of the shift provides an average sub-frame offset between arrival and depletion times.
In further embodiments of the present invention the calculation of the shift in the play-out of the data packets is implemented by first calculating the shift to address the frame phase offset and then calculating the watermark optimization or alternatively implementing a combined shift calculated by summing the shifts due to the frame phase offset and watermark optimization.
The present invention also provides a communication system receiver for receiving a transported stream of data packets over a packet switched channel through a LAN. The receiver includes a de-packetizer which receives the packets received off the LAN and strips off the header of each packet. The receiver further includes a jitter buffer that receives the payload, which includes one or more frames and a sequence number. The jitter buffer stores the frames according to arrival times and plays out the frames in an order defined by the sequence numbers. A controller for the jitter buffer controls the order and timing of the play-out of the data frames. The receiver includes a decoder which receives frames output from the jitter buffer at regular intervals. The decoder reconstructs the original signal. In a preferred embodiment, the receiver of the present invention includes a first clock having a first rate that defines arrival sampling times which runs at a rate faster than a second clock having a second rate that defines frame depletion times.
The foregoing features and advantages of the invention will be apparent from the following more particular description of preferred embodiments of the invention, as illustrated in the accompanying drawings in which like reference characters refer to the same parts throughout the different views. The drawings are not necessarily to scale, emphasis instead being placed upon illustrating the principles of the invention.